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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
Autres articles (81)
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.
Sur d’autres sites (11317)
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libavformat/rtpdec : Fix RTP timestamp wraparound in Producer Reference Time
7 juin, par Clément Péronlibavformat/rtpdec : Fix RTP timestamp wraparound in Producer Reference Time
The rtp_set_prft() function incorrectly calculates the timestamp delta
when RTP timestamps wrap around the 32-bit boundary. The current code :delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp ;
treats both timestamps as large positive values, causing wraparound to
produce a large negative delta instead of the correct small positive delta.For example, with a 90kHz video clock :
last_rtcp_timestamp = 0xFFFFFF00 (near wraparound)
timestamp = 0x00000100 (after wraparound)
Current result : delta ≈ -4.3 billion ticks ≈ -47,721 seconds
Expected result : delta ≈ +512 ticks ≈ +0.006 seconds
This causes prft->wallclock to jump backward by approximately :
90kHz video : 47,721 seconds ( 13.25 hours)
48kHz audio : 89,478 seconds ( 24.9 hours)
8kHz audio : 536,871 seconds ( 6.2 days)
Fix by casting the subtraction result to int32_t, which correctly
handles wraparound through modular arithmetic :delta_timestamp = (int32_t)(timestamp - s->last_rtcp_timestamp) ;
This ensures the delta is always in the range [-2^31, 2^31-1], making
wraparound produce the correct small positive values.Fixes timing jumps in applications that rely on Producer Reference Time
for media synchronization.Signed-off-by : Clément Péron <peron.clem@gmail.com>
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc> -
Revision 33150 : Essayer de suivre les optimisations de la 2.0.10++ SVN
23 novembre 2009, par real3t@… — LogEssayer de suivre les optimisations de la 2.0.10++ SVN
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webrtc to rtmp send video from camera to rtmp link
14 avril 2024, par Leo-Mahendrai cant send the video from webrtc which is converted to bufferd data for every 10seconds and send to server.js where it takes it via websockets and convert it to flv format using ffmpeg.


i am trying to send it to rtmp server named restreamer for start, here i tried to convert the buffer data and send it to rtmp link using ffmpeg commands, where i initially started to suceesfully save the file from webrtc to mp4 format for a duration of 2-3 minute.


after i tried to use webrtc to send video data for every 10 seconds and in server i tried to send it to rtmp but i cant send it, but i can see the connection of rtmp url and server is been taken place but i cant see the video i can see the logs in rtmp server as


2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37700" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH"
2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37716" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH"
2024-04-14 12:35:45 ts=2024-04-14T07:05:45Z level=INFO component="RTMP" msg="no streams available" action="INVALID" address=":1935" client="172.17.0.1:37728" path="/3d30c5a9-2059-4843-8957-da963c7bc19b.stream" who="PUBLISH" 



my frontend code


const handleSendVideo = async () => {
 console.log("start");
 
 if (!ws) {
 console.error('WebSocket connection not established.');
 return;
 }
 
 try {
 const videoStream = await navigator.mediaDevices.getUserMedia({ video: true });
 const mediaRecorder = new MediaRecorder(videoStream);
 
 const requiredFrameSize = 460800;
 const frameDuration = 10 * 1000; // 10 seconds in milliseconds
 
 mediaRecorder.ondataavailable = async (event) => {
 if (ws.readyState !== WebSocket.OPEN) {
 console.error('WebSocket connection is not open.');
 return;
 }
 
 if (event.data.size > 0) {
 const arrayBuffer = await event.data.arrayBuffer();
 const uint8Array = new Uint8Array(arrayBuffer);
 
 const width = videoStream.getVideoTracks()[0].getSettings().width;
 const height = videoStream.getVideoTracks()[0].getSettings().height;
 
 const numFrames = Math.ceil(uint8Array.length / requiredFrameSize);
 
 for (let i = 0; i < numFrames; i++) {
 const start = i * requiredFrameSize;
 const end = Math.min((i + 1) * requiredFrameSize, uint8Array.length);
 let frameData = uint8Array.subarray(start, end);
 
 // Pad or trim the frameData to match the required size
 if (frameData.length < requiredFrameSize) {
 // Pad with zeros to reach the required size
 const paddedData = new Uint8Array(requiredFrameSize);
 paddedData.set(frameData, 0);
 frameData = paddedData;
 } else if (frameData.length > requiredFrameSize) {
 // Trim to match the required size
 frameData = frameData.subarray(0, requiredFrameSize);
 }
 
 const dataToSend = {
 buffer: Array.from(frameData), // Convert Uint8Array to array of numbers
 width: width,
 height: height,
 pixelFormat: 'yuv420p',
 mode: 'SendRtmp'
 };
 
 console.log("Sending frame:", i);
 ws.send(JSON.stringify(dataToSend));
 }
 }
 };
 
 // Start recording and send data every 10 seconds
 mediaRecorder.start(frameDuration);
 
 console.log("MediaRecorder started.");
 } catch (error) {
 console.error('Error accessing media devices or starting recorder:', error);
 }
 };



and my backend


wss.on('connection', (ws) => {
 console.log('WebSocket connection established.');

 ws.on('message', async (data) => {
 try {
 const parsedData = JSON.parse(data);

 if (parsedData.mode === 'SendRtmp' && Array.isArray(parsedData.buffer)) {
 const { buffer, pixelFormat, width, height } = parsedData;
 const bufferArray = Buffer.from(buffer);

 await sendRtmpVideo(bufferArray, pixelFormat, width, height);
 } else {
 console.log('Received unknown or invalid mode or buffer data');
 }
 } catch (error) {
 console.error('Error parsing WebSocket message:', error);
 }
 });

 ws.on('close', () => {
 console.log('WebSocket connection closed.');
 });
 });
 const sendRtmpVideo = async (frameBuffer, pixelFormat, width, height) => {
 console.log("ffmpeg data",frameBuffer)
 try {
 const ratio = `${width}x${height}`;
 const ffmpegCommand = [
 '-re',
 '-f', 'rawvideo',
 '-pix_fmt', pixelFormat,
 '-s', ratio,
 '-i', 'pipe:0',
 '-c:v', 'libx264',
 '-preset', 'fast', // Specify the preset for libx264
 '-b:v', '3000k', // Specify the video bitrate
 '-loglevel', 'debug',
 '-f', 'flv',
 // '-flvflags', 'no_duration_filesize', 
 RTMPLINK
 ];


 const ffmpeg = spawn('ffmpeg', ffmpegCommand);

 ffmpeg.on('exit', (code, signal) => {
 if (code === 0) {
 console.log('FFmpeg process exited successfully.');
 } else {
 console.error(`FFmpeg process exited with code ${code} and signal ${signal}`);
 }
 });

 ffmpeg.on('error', (error) => {
 console.error('FFmpeg spawn error:', error);
 });

 ffmpeg.stderr.on('data', (data) => {
 console.error(`FFmpeg stderr: ${data}`);
 });

 ffmpeg.stdin.write(frameBuffer, (err) => {
 if (err) {
 console.error('Error writing to FFmpeg stdin:', err);
 } else {
 console.log('Data written to FFmpeg stdin successfully.');
 }
 ffmpeg.stdin.end(); // Close stdin after writing the buffer
 });
 } catch (error) {
 console.error('Error in sendRtmpVideo:', error);
 }
 };